On this link you find necessary info on resetten a IP phone
https://www.ucguru.com/reset-cisco-phone-factory-defaults/
But sometimes you brick a phone. I had some troubles with phone having firmware pre 8.5.2. After hard resetting a few phones I was left a few blank bricks.
The cure ? A Callmanager Express with firmware 8.5.2.
1. Enable CME (telephony-service etc.)
2. Add firmware to flash and config
3. Enable autoregistration or add ephones with the MAC addresses of the phones
4. Enable DHCP
5. Connect phones
As soon as the phones get a config XML from the CME, life will flow back into the phones. Firmware will be loaded and screens will pop on.
Friday, 23 January 2015
SIP-trunking with Routit (Broadsoft/Broadworks) part 3
In the first 2 parts I showed you how registration with Routit works and how to avoid problems with the global SIP-proxy statement. In this blog I add some extra info and knowledge obtained by implementing Routit SIP-trunks.
Infrastructure
First get a IP-VPN. Only fools buys and bastards sell Internet based SIP trunks. Sorry sales it will add a extra 30 euro's and destroys your case maybe but it will pay off in the end.
Config an IP VPN (1x /24 segment), with the CPE-router having an address in the Voice VLAN of the customer. Only for multisegment customers you need an IP VPN Plus (costomers with multiple locations/branches with different IP-segements).
Internet based connection
If you go for an Internet based (#@$) SIP-trunk do the following: Use a modern firewall/router in front of the PBX. It should be able to do SIP Inspect (SIP ALG). Do not connect the Internet connection directly to the UC or CUBE. It will make troubleshooting troublesome because it is not always clear which source address it uses in it's SIP packets. It will work in the end but I tend to avoid this. Next to that just do not combine edge security and voip on your voice gateway/PBX.
I assume this setup:
This setup has a few caveats:
Outbound calls can only have the main number as calling party
Inbound calls do not get the DDI in the request URI of the INVITE, only in the To: header. Routit always sends the registration number in the INVITE request URI
There are 3 workarounds:
In all cases do some header stripping because during any support call they will tell you they see unnecessary header info. Add the following:
For outbound calls be sure to replace all calling party references with the registration number. This is quite easy with a translation profile
I will add another blog for getting things working with CUBE+CUCM (and IP-VPN!).
Infrastructure
First get a IP-VPN. Only fools buys and bastards sell Internet based SIP trunks. Sorry sales it will add a extra 30 euro's and destroys your case maybe but it will pay off in the end.
Config an IP VPN (1x /24 segment), with the CPE-router having an address in the Voice VLAN of the customer. Only for multisegment customers you need an IP VPN Plus (costomers with multiple locations/branches with different IP-segements).
Internet based connection
If you go for an Internet based (#@$) SIP-trunk do the following: Use a modern firewall/router in front of the PBX. It should be able to do SIP Inspect (SIP ALG). Do not connect the Internet connection directly to the UC or CUBE. It will make troubleshooting troublesome because it is not always clear which source address it uses in it's SIP packets. It will work in the end but I tend to avoid this. Next to that just do not combine edge security and voip on your voice gateway/PBX.
I assume this setup:
Outbound calls can only have the main number as calling party
Inbound calls do not get the DDI in the request URI of the INVITE, only in the To: header. Routit always sends the registration number in the INVITE request URI
There are 3 workarounds:
- Use a TCL-script to get the info from the To: field to the internal DNIS field (works) link
- Use a SIP-profile to change inbound SIP-headers (not tested by me) link link2
- Add the DDI to the extension as secondary number (UC500/CME) works.
In all cases do some header stripping because during any support call they will tell you they see unnecessary header info. Add the following:
Voice service voipDo not set asserted-id under voice service voip. It will not work when connected via the Internet.
sip
sip-profiles 1000
!
voice class sip-profiles 1000
request ANY sip-header Cisco-Guid remove
response ANY sip-header Cisco-Guid remove
request ANY sdp-header Connection-Info remove
response ANY sdp-header Connection-Info remove
!
For outbound calls be sure to replace all calling party references with the registration number. This is quite easy with a translation profile
translation-profile outboundYou might add the following when using a CUBE:
translate calling 1
!
translation rule 1
rule 15 /.*/ /31XXXXXXXXX/
!
voice class sip-profiles 1000
request INVITE sip-header P-Asserted-Identity add "P-Asserted-Identity:<sip:registratienummer@xxx.yyy.voipit.nl>"
!
I will add another blog for getting things working with CUBE+CUCM (and IP-VPN!).
Wednesday, 21 January 2015
Cisco announces Multigigabit switches
To support the higher throughput o 802.11ac wave 2. Cisco bundled forces with other vendors to develop NBaseT, a standard that enables 2.5 and 5 Gigabit/s on Cat 5e/6 cabling. Now the first batch of switches are announced (pre-standard that is).
Traceroute
An interesting article on traceroute
https://rawhex.com/2015/01/an-unhealthy-journey-into-the-world-of-the-traceroute/
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